[asterisk-users] Re: Polycom 601 loop

[Nathan Bell]

This is what I get from the asterisk CLI:

ast*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status   
202                        (Unspecified)    D          0        Unmonitored
201                        (Unspecified)    D          0        Unmonitored
2 sip peers [2 online , 0 offline]
ast*CLI> sip show users
Username                   Secret           Accountcode      
Def.Context      ACL  NAT      
202                        *******                   from-sip         
No   RFC3581  
201                        *******                   from-sip         
No   RFC3581  
ast*CLI>

dave cantera wrote:

> and
> sip show users
>
>
> Noah Miller wrote:
>
>> Hi Nathan -
>>
>>> No loop now, but instead I get this:
>>>
>>> Mar 26 15:42:18 NOTICE[1854] app_dial.c: Unable to create channel of
>>> type 'SIP' (cause 3 - No route to destination)
>>> Mar 26 15:42:18 VERBOSE[1854] logger.c:   == Everyone is busy/congested
>>> at this time (1:0/0/1)
>>> Mar 26 15:42:18 DEBUG[1854] app_dial.c: Exiting with 
>>> DIALSTATUS=CHANUNAVAIL.
>>
>>
>> Is the SIP device at 201 registered?  What happens when you do a "sip
>> show peers"?
>>
>> - Noah
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>>
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